This is a trivial but dangerous change to remove X11/Xlib.h out of
shared_x_display.h. Since we do not have a strict and automatically
Include-What-You-Use rule, I cannot quite tell whether any Chromium source files
wrongly assume X11/Xlib.h will be included through shared_x_display.h. We can
fix the breaks in Chromium after this change has been failed to integrate to
Chromium.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2482963003
Cr-Commit-Position: refs/heads/master@{#14987}
Reason for revert:
Broke dependent project because the .gn changes weren't accompanied by corresponding .gyp changes.
Original issue's description:
> Add a webrtc{en,de}coderfactory implementation for VideoToolbox
>
> This CL removes the coupling of the VideoToolbox h264 implementation
> to the generic h264 code. The files have been moved into sdb/obj/Framework
> and all dependency on them has been removed from the rest of WebRTC.
> We now add it as an external encoder via a factory supplied to the
> CreatePeerConnectionFactory call. This also brings the iOS implementation
> closer to what we do on Android for MediaCodec.
>
> BUG=webrtc:6619
>
> Committed: https://crrev.com/6a5047dad31f14f53dd9f8bc1ecde19e1dede2e4
> Cr-Commit-Position: refs/heads/master@{#14953}
TBR=magjed@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org
BUG=webrtc:6619
Review-Url: https://codereview.webrtc.org/2483273002
Cr-Commit-Position: refs/heads/master@{#14985}
The new format for plot lines is PLOT <plot_no> <var_name>:<ssrc>@<alg_name> <time> <value>. The var_name is no longer prefixed by the context/tag (which most of the time was just the same as the test name.)
Update plot_dynamics.py script (which didn't work) to visualize the new BWE_TEST_LOGGING_PLOT lines.
BUG=webrtc:6621
Review-Url: https://codereview.webrtc.org/2456373002
Cr-Commit-Position: refs/heads/master@{#14983}
This change changes mixing to be done at the lowest possible
APM-native rate that does not lead to quality loss. An Audio
Processing-native rate is one of 8, 16, 32, or 48 kHz. Mixing at a
lower sampling rate and avoiding resampling can in many cases lead to
big efficiency improvements, as reported by experiments.
This CL also fixes a design issue with the AudioMixer: audio at
non-native rates is no longer fed to the APM instance which is the
limiter.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2458703002
Cr-Commit-Position: refs/heads/master@{#14980}
Not all members are collected by RTCStatsCollector and detached tracks
are not visible in the returned stats. This needs to be addressed before
closing crbug.com/660827 and crbug.com/659137
BUG=chromium:660827, chromium:659137, chromium:627816
Review-Url: https://codereview.webrtc.org/2467873005
Cr-Commit-Position: refs/heads/master@{#14978}
git cl try will then trigger CQ dry run, achieving even better
result, because PRESUBMIT will be run as well.
TBR=tandrii@chromium.org
BUG=663320
NOTRY=True
Review-Url: https://codereview.webrtc.org/2489523003
Cr-Commit-Position: refs/heads/master@{#14976}
I have no idea how bad it is that we're hitting this limit; I'm just
doing this to stop the tests from flaking.
BUG=webrtc:6484
Review-Url: https://codereview.webrtc.org/2477663002
Cr-Commit-Position: refs/heads/master@{#14974}
The audio_device_module field was currently unused. The audio_mixer
field is going to be used to pass an AudioMixer to AudioState.
In the hopefully-not-very-far future, the toplevel WebRTC API will allow passing
a custom AudioMixer, e.g. for spatialized audio (audio in space). If no
mixer is passed, a default mixer is created (the one in modules/audio_mixer).
The only object which will have a permanent reference to the mixer is AudioState.
AudioState is created in WebRTCVoiceEngine with a configuration object,
which already contains a VoiceEngine pointer. In this CL, we extend this
config object with a mixer pointer.
In summary: in an upcoming CL, a mixer will be either created in or passed to
WebRTCVoiceEngine. This mixer will be passed to the ctor of AudioState in a
config struct.
BUG=webrtc:6346
NOTRY=True
Review-Url: https://codereview.webrtc.org/2456363002
Cr-Commit-Position: refs/heads/master@{#14973}
Reason for revert:
The usage of UIKeyboardTypeASCIICapableNumberPad enum (available only for iOS 10.0), is breaking the build of AppRTCMobile on devices with lesser os.
To re-land the UIKeyboardTypeASCIICapableNumberPad should be replaced with UIKeyboardTypeNumberPad
Original issue's description:
> Add bitrate section to settings view controller.
>
> BUG=webrtc:6654
>
> Committed: https://crrev.com/3babb99039478c36be58171c5409eac07ae153e5
> Cr-Commit-Position: refs/heads/master@{#14952}
TBR=magjed@webrtc.org,kthelgasson@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6654
Review-Url: https://codereview.webrtc.org/2488653002
Cr-Commit-Position: refs/heads/master@{#14972}
The BitrateEstimatorTest contains code to initialize an AudioState
instance and an AudioReceiveStream. That code is never run, and is
deleted in this CL.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2479383003
Cr-Commit-Position: refs/heads/master@{#14971}
Moving one of the targets that shouldn't pollute the already large
webrtc/BUILD.gn file.
BUG=webrtc:6440
NOTRY=True
Review-Url: https://codereview.webrtc.org/2440963002
Cr-Commit-Position: refs/heads/master@{#14970}
As I was preparing to move some files from the api/ folder, I noticed
that this file was not included in the BUILD.gn file. I've added it back
in and updated it to compile and run successfully again.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2485603002
Cr-Commit-Position: refs/heads/master@{#14965}
This CL removes WebRtcVideoReceiveStream::FilterSupportedCodecs and
adds more fine grained code for selecting and verifying codecs. This
also removes unnecessary copying of cricket::VideoCodecs.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2474433012
Cr-Commit-Position: refs/heads/master@{#14964}
D3D_FEATURE_LEVEL_11_0 is not offically documented on MSDN to be a requirement
of DXGI duplicator APIs. So instead of using D3D_FEATURE_LEVEL_11_0 as a
requirement in D3dDevice::Initialize(), this change adds a
ScreenCapturerWinDirectx::SupportedFeatureLevel() function to retrieves minimum
and maximium for further reference (in HostTraits to control the experiment).
BUG=314516
Review-Url: https://codereview.webrtc.org/2468083002
Cr-Commit-Position: refs/heads/master@{#14962}
This change replaces all GetWindowList / GetScreenList with GetScreenList,
SelectWindow / SelectScreen with SelectSource, and BringSelectedWindowToFront
with FocusOnSelectedSource in WebRTC.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2479553006
Cr-Commit-Position: refs/heads/master@{#14960}
Half of the function's code was reformatted and reindented in
https://codereview.webrtc.org/1307663004, but the bottom half was still
adhering to an old coding style and using different indentation values.
Not only does this make the code look confusing, but it can cause build
issues on certain compilers: for example, GCC 6.2.0 with -Wall causes
the build to fail because -Wmisleading-indentation is enabled.
BUG=None
R=asapersson@webrtc.org,danilchap@webrtc.org
Review-Url: https://codereview.webrtc.org/2479193002
Cr-Commit-Position: refs/heads/master@{#14957}
The cricket::Codec class contains std containers like
std::map<std::string, std::string> and is expensive to copy. This CL
adds move constructors and move assignment operators for it and all
subclasses.
This CL also:
* Implement functions with '= default' instead of doing it manually.
* Makes codec::Matches symmetric. We currently don't check if the
payload type of the callee is static, and might incorrectly return
true in these cases.
BUG=None
Review-Url: https://codereview.webrtc.org/2481193002
Cr-Commit-Position: refs/heads/master@{#14956}
The unit was kbps but the one default use of it is in bps. The inconsistency should be fixed.
BUG=webrtc:6670
Review-Url: https://codereview.webrtc.org/2247213005
Cr-Commit-Position: refs/heads/master@{#14955}
This CL removes the coupling of the VideoToolbox h264 implementation
to the generic h264 code. The files have been moved into sdb/obj/Framework
and all dependency on them has been removed from the rest of WebRTC.
We now add it as an external encoder via a factory supplied to the
CreatePeerConnectionFactory call. This also brings the iOS implementation
closer to what we do on Android for MediaCodec.
BUG=webrtc:6619
Review-Url: https://codereview.webrtc.org/2463313002
Cr-Commit-Position: refs/heads/master@{#14953}
- Removes the lock that was used to protect the audio transport object.
It is now protected "by design" instead.
- Removes rec/play_bytes_per_sample_ since we only support 16-bit samples.
BUG=webrtc:6560
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/2466613002 .
Cr-Commit-Position: refs/heads/master@{#14950}
This change the interface of the PacketBuffer since the bitstream data of the packet has to be persistent when inserted into the PacketBuffer.
BUG=webrtc:5514
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2476283002 .
Cr-Commit-Position: refs/heads/master@{#14949}
To differentiate between when a packet should be dropped or when a keyframe
is missing its SPS and/or SPS CopyAndFixBitstream now returns:
- kInsert, the packet should be inserted into the PacketBuffer.
- kDrop, the packet should be dropped.
- kRequestKeyframe, a keyframe should be requested.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2477343002
Cr-Commit-Position: refs/heads/master@{#14948}
- Change const ptr to const ref in parameter list.
Using nullptr as argument was invalid, so no need to send
pointer instead of reference.
- Change return type to void or bool, where appropriate
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2455963003
Cr-Commit-Position: refs/heads/master@{#14945}
Prior to this change, we signalled that ULPFEC was disabled
through a bool, but that RED was disabled by setting its
payload type to -1. The latter is consistent with how we
disable RED/ULPFEC in the config, so this CL removes the
ULPFEC bool from the {,Set}UlpfecConfig chain of member
functions.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2460533002
Cr-Commit-Position: refs/heads/master@{#14944}
At the same time, change to using int's instead of uint8_t's for the payload type.
This allows us to signal disabled FEC or RED using the sentinel value -1, which
is commonplace in other parts of the code.
These APIs will be deprecated when ULPFEC is deprecated.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2448463003
Cr-Commit-Position: refs/heads/master@{#14942}
Since ffmpeg use an optimized bitstream reader that reads in chunks of 32/64
bits the bitstream buffer has to be increased in order for the reader to not
read out of bounds.
BUG=webrtc:5514
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2476513004 .
Cr-Commit-Position: refs/heads/master@{#14941}
Also improve denoiser to not assume identical stride of all involved
frames, and delete the no longer needed function I420Buffer::CopyKeepStride.
BUG=None
Review-Url: https://codereview.webrtc.org/2469763002
Cr-Commit-Position: refs/heads/master@{#14940}
